Convert TS to DSS — Free Online Tool
Convert MPEG-2 Transport Stream (.ts) files to Digital Speech Standard (.dss) format using the ADPCM IMA OKI codec — extracting and compressing the audio track into a format designed for Olympus, Philips, and Grundig digital dictation devices. This tool runs entirely in your browser with no file uploads required.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your TS file here
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Settings
Note: Browser-based encoding uses approximate quality targets. For precise CRF compression, copy the FFmpeg command above and run it on your desktop.
Estimated output:
Conversion Complete!
DownloadHow It Works
TS files are broadcast containers that can carry multiple audio tracks (AAC, MP3, AC3, Opus, FLAC) alongside video and subtitle streams. During this conversion, all video streams and subtitles are discarded entirely — DSS is a audio-only format with no video support. The selected audio track is decoded from its source codec and then re-encoded using ADPCM IMA OKI, a lossy adaptive delta pulse-code modulation algorithm developed specifically for the DSS format. ADPCM IMA OKI operates at a very low bitrate optimized for speech intelligibility rather than music fidelity, so the resulting file will be significantly smaller than the original TS audio but with notably reduced audio quality. If your TS file contains multiple audio tracks, FFmpeg will select the first audio stream by default.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program — the open-source multimedia processing engine that powers this conversion both in the browser (via FFmpeg.wasm) and on the desktop. |
-i input.ts
|
Specifies the input file — an MPEG-2 Transport Stream, which may contain H.264 or H.265 video, one or more audio tracks (AAC, AC3, MP3, etc.), subtitles, and broadcast metadata. FFmpeg reads and demuxes all of these streams before processing. |
-c:a adpcm_ima_oki
|
Sets the audio codec to ADPCM IMA OKI — the specific low-bitrate adaptive delta PCM encoder required by the DSS format. This re-encodes the TS audio (whatever its original codec) into the speech-optimized compressed format that Olympus/Philips/Grundig dictation devices expect. |
output.dss
|
Defines the output file with a .dss extension, which tells FFmpeg to write a Digital Speech Standard container. Since DSS is audio-only, FFmpeg automatically drops the video and subtitle streams from the TS input — no explicit flag to discard video is needed because the output format cannot contain it. |
Common Use Cases
- Archiving spoken-word content from recorded broadcast television segments onto a digital dictation device for playback on an Olympus or Philips recorder
- Extracting commentary or narration from a broadcast TS recording to import into dictation transcription software that requires DSS input
- Converting recorded interview segments captured via a TV tuner card (saved as .ts) into DSS for use with voice recognition or transcription workflows
- Preparing audio from broadcast news clips for archival in legacy digital dictation systems used in legal or medical documentation environments
- Stripping and compressing the audio from a live-stream recording (.ts from HLS capture) into a compact DSS file for storage on dictation hardware with limited memory
Frequently Asked Questions
Likely not at the same fidelity level. DSS was designed specifically for speech intelligibility on dictation devices, and the ADPCM IMA OKI codec it uses operates at a very low bitrate. If your TS file contains broadcast-quality AAC or AC3 audio at 192k or higher, you will hear a noticeable drop in quality — background music, sound effects, and complex audio will sound degraded. Spoken voice content fares significantly better, which is exactly what the format was engineered for.
The video stream is completely dropped. DSS is an audio-only format — it has no provision for video data whatsoever. The FFmpeg command targets only the audio stream (`-c:a adpcm_ima_oki`) and the output container (.dss) cannot hold video. Only the first audio track from your TS file will be included in the output by default.
By default, FFmpeg selects the first audio stream in the TS file. TS containers commonly carry multiple audio tracks for multilingual broadcast content. If you need a specific track — for example, the second language stream — you would modify the command locally by adding `-map 0:a:1` (for the second audio track, zero-indexed) before the output filename. DSS supports only a single audio track, so only one stream can be written to the output.
No — the DSS format with the ADPCM IMA OKI codec does not expose configurable bitrate or quality parameters in FFmpeg. The bitrate and encoding characteristics are fixed by the codec specification itself. This is unlike the TS input format, which supports flexible quality settings. You get what the codec defines, which is a low-bitrate, speech-optimized output.
You can use a shell loop to process multiple files. On Linux or macOS, run: `for f in *.ts; do ffmpeg -i "$f" -c:a adpcm_ima_oki "${f%.ts}.dss"; done`. On Windows Command Prompt, use: `for %f in (*.ts) do ffmpeg -i "%f" -c:a adpcm_ima_oki "%~nf.dss"`. This browser-based tool processes one file at a time, so the desktop FFmpeg command is especially useful for bulk conversions.
No. DSS does not support subtitles, chapter markers, or rich metadata. TS files from broadcast sources often contain EIT metadata, teletext subtitles, and multiple service streams — all of this is discarded during conversion. The DSS output will contain only the re-encoded audio. If preserving subtitles or broadcast metadata is important, DSS is the wrong target format for that purpose.
Technical Notes
The DSS (Digital Speech Standard) format is a niche, proprietary container developed jointly by Olympus, Philips, and Grundig, and its FFmpeg support is limited to the `adpcm_ima_oki` codec — a variant of IMA ADPCM tuned for the low-bitrate speech characteristics of the DSS specification. Unlike the TS input, which supports a wide range of video and audio codecs including H.264, H.265, VP9, AAC, AC3, FLAC, and Opus, DSS is entirely audio-only and offers no quality configuration knobs through FFmpeg. The sample rate of the output will be constrained by what the ADPCM IMA OKI codec supports (typically 8000 Hz), meaning any higher-quality broadcast audio in your TS file will be downsampled significantly. File sizes will be dramatically smaller than the original TS, but this comes at the cost of audio bandwidth. DSS files produced by FFmpeg may not be fully compatible with all dictation hardware or proprietary Olympus/Philips software, as those ecosystems sometimes use slightly variant DSS implementations. Testing playback on your target device before bulk conversion is recommended.